The MPEG audio coding standard is the international standard for the compression of digital audio signals. It can be applied both for audiovisual and audio-only applications to significantly reduce the requirements of transmission bandwidth and data storage with low distortion. The second phase of MPEG, labeled as MPEG-2, aims to support all the normative feature listed in MPEG-1 audio and provide extension capabilities of multi-channel and multilingual audio on an extension of standard to lower sampling frequencies and lower bit rates. No matter what is MPEG-1 or MPEG-2 standard, the MPEG audio compression standard defines threes layers of compression, named as Layer I, II, and III. Each successive layer offers better compression performance, but at a higher complexity and computation cost. Layer I and II are basically similar and based on subband coding. The difference between them mainly lies in formatting side information and a finer quantization is provided in Layer II. Layer III adopts more complex schemes such as hybrid filterbank, Huffman coding and non-linear quantization. From the viewpoint of hardware complexity and achieved quality, Layer II might be a reasonable compromise for general usage. In the official ISO/MPEG subject tests, Layer II coding shows an excellent performance of CD quality at a 128 Kbps per monophonic channel.
The MPEG-2 audio coding standard is an extension of the MPEG-1. With backward compatibility, it is possible to produce a multi-channel audio at any time without making the two-channel MPEG-1 obsolete. In MPEG-2 audio coding, five audio channels L (left), R (right), C (central), LS (left surround), RS (right surround) are mapped to five transmission channels T0, T1, T2, T3, T4. The T0 and T1 channels are compatible with the left and right basic transmission channels of MPEG-1 audio, and the T2 to T4 channels are extended transmission channels. Consequently, in MPEG-2 audio decoding a multichannel decoder is required to reconstruct multichannel audio signals.
The MPEG-2 audio decoding related to inverse quantization and multichannel processing is described in FIG. 1. The coded data is subjected to frame unpacking while decoding of bit allocation and scale factor. The quantized data is recovered through an operation of inverse quantization, and reconstructed data can be generated through multichannel processing followed by synthesizing 32 subband data in a subband synthesis filter.
The elementary concept behind MPEG is based on the multirate subband-based coding techniques. Basically, the most computational load depends highly on the realization of the synthesis subband in the decoder, and can be reduced using the regular fast algorithm, such as inverse-modified discrete cosine transform (IMDCT) and fast Fourier transform (FET) with data shifting and rearrangement. As for the other important computational parts of the decoding, inverse quantization (IQ) and multichannel processing (MC) are seldom mentioned and seem to be unsuitable when applying a fast algorithm based on the characteristics of complex control and irregular data flow.
Referring to the architecture design, different aspects of the architecture must be utilized in the MPEG-2 audio decoder. These designs are basically applied either as general purpose DSP-based techniques such as stand-alone chip sets, or proposed as architecture dedicated to the MPEG-2 audio bitstream. Whether the architecture is DSP-based or is dedicated architecture, most processors implement the MPEG-2 decoding by programming. However, these processors suffer from considerable overheads of computation and control. Moreover, some papers have only focused on the synthesis subband with a dedicated cost-effective architecture. In that case, they must perform the IQ and MC in the host platform, such as PC. These designs also increase the complexity in the interface and communication between the dedicated chip sets and the host.